diff options
author | file <file@f38db490-d61c-443f-a65b-d21fe96a405b> | 2009-11-06 16:56:37 +0000 |
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committer | file <file@f38db490-d61c-443f-a65b-d21fe96a405b> | 2009-11-06 16:56:37 +0000 |
commit | 200c0b5636015b0bb4846939633d14b42e710301 (patch) | |
tree | 2d2ad22aff2deedc4b1e98f0a2cc907e711c6897 | |
parent | cd582b00b7b447a2783942cb457a3cf85225b3b3 (diff) |
Fix a crash caused by freeing a dialog directly instead of using dialog_unref.
(closes issue #16097)
Reported by: steinwej
Patches:
no_RTP.diff uploaded by steinwej (license 841)
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@228415 f38db490-d61c-443f-a65b-d21fe96a405b
-rw-r--r-- | channels/chan_sip.c | 21 |
1 files changed, 3 insertions, 18 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c index 70e2b7d0c..9ac215491 100644 --- a/channels/chan_sip.c +++ b/channels/chan_sip.c @@ -6218,7 +6218,7 @@ static struct sip_pvt *sip_alloc(ast_string_field callid, struct sockaddr_in *si return NULL; if (ast_string_field_init(p, 512)) { - ast_free(p); + dialog_unref(p); return NULL; } @@ -6278,27 +6278,12 @@ static struct sip_pvt *sip_alloc(ast_string_field callid, struct sockaddr_in *si p->udptl = ast_udptl_new_with_bindaddr(sched, io, 0, bindaddr.sin_addr); p->t38_maxdatagram = global_t38_maxdatagram; } - if (!p->rtp|| (ast_test_flag(&p->flags[1], SIP_PAGE2_VIDEOSUPPORT) && !p->vrtp) + if (p->rtp|| (ast_test_flag(&p->flags[1], SIP_PAGE2_VIDEOSUPPORT) && !p->vrtp) || (ast_test_flag(&p->flags[1], SIP_PAGE2_TEXTSUPPORT) && !p->trtp)) { ast_log(LOG_WARNING, "Unable to create RTP audio %s%ssession: %s\n", ast_test_flag(&p->flags[1], SIP_PAGE2_VIDEOSUPPORT) ? "and video " : "", ast_test_flag(&p->flags[1], SIP_PAGE2_TEXTSUPPORT) ? "and text " : "", strerror(errno)); - if (p->rtp) { - ast_rtp_destroy(p->rtp); - } - if (p->vrtp) { - ast_rtp_destroy(p->vrtp); - } - if (p->udptl) { - ast_udptl_destroy(p->udptl); - } - ast_mutex_destroy(&p->pvt_lock); - if (p->chanvars) { - ast_variables_destroy(p->chanvars); - p->chanvars = NULL; - } - ast_string_field_free_memory(p); - ast_free(p); + dialog_unref(p); return NULL; } ast_rtp_setqos(p->rtp, global_tos_audio, global_cos_audio, "SIP RTP"); |